[sip-comm-dev] Conference Call Codec negotiation issue


#1

Hi

I would like to report an issue on the conference call using SIP, The User A make a SIP call to user B, then after the call is established negotiating the codec g722 or Speex, the User A add a PSTN Callee into the the call. The User A and PSTN GW negotiate a new codec (PCMA or PCMU) but no re-negotiation is sent to User B that stay into the conference with g722 or Speex producing "Martians" audio. When the User A ,that is the one added a new party into the call, renegotiate the codec, should then re-invite all the other parties to renegotiate the new codec.

The problem can be reproduced all the time.

Thanks for the attention

Fabio Galdi

···

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#2

Hey Fabio,

На 08.12.10 16:38, Fabio Telme написа:

Hi

I would like to report an issue on the conference call using SIP,
The User A make a SIP call to user B, then after the call is
established negotiating the codec g722 or Speex, the User A add a
PSTN Callee into the the call. The User A and PSTN GW negotiate a
new codec (PCMA or PCMU) but no re-negotiation is sent to User B
that stay into the conference with g722 or Speex producing
"Martians" audio. When the User A ,that is the one added a new
party into the call, renegotiate the codec, should then re-invite
all the other parties to renegotiate the new codec.

Why would you want to do that?

SIP Communicator handles the transcoding.

This way whoever supports wideband can keep on using it and benefit from
good quality, and for everyone else it's just the same.

Can you explain your thoughts?

Emil

···

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#3

Hi Emil,

Thanks for the answer, i did not know that SC is able to transcode, that's really good, so i think the issue should be addressed some where else,
the problem i face on the conference call is that when i introduce a party in the conference that is on PSTN trunk, the audio is good only from the user that has added the PSTN party into the conference, all the other parties cannot get and produce good audio, there is only a noisy sound.
So here the example :

User A call User B, then after call estabilished User A add a PSTN user into the conference on Mobile phone or landline. The 3 party are in conference but only User A and PSTN has good Audio, the User B receive only noisy audio.
The same test can be done with User B add the PSTN number into the conference, and in this case User B and PSTN are fine and User A got noisy audio.

hope some one could reproduce the same issue.

Regards
Fabio

···

Il giorno 08/dic/2010, alle ore 19.32, Emil Ivov ha scritto:

Hey Fabio,

На 08.12.10 16:38, Fabio Telme написа:

Hi

I would like to report an issue on the conference call using SIP,
The User A make a SIP call to user B, then after the call is
established negotiating the codec g722 or Speex, the User A add a
PSTN Callee into the the call. The User A and PSTN GW negotiate a
new codec (PCMA or PCMU) but no re-negotiation is sent to User B
that stay into the conference with g722 or Speex producing
"Martians" audio. When the User A ,that is the one added a new
party into the call, renegotiate the codec, should then re-invite
all the other parties to renegotiate the new codec.

Why would you want to do that?

SIP Communicator handles the transcoding.

This way whoever supports wideband can keep on using it and benefit from
good quality, and for everyone else it's just the same.

Can you explain your thoughts?

Emil

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#4

Hi Emil,

Thanks for the answer, i did not know that SC is able to transcode, that's really good, so i think the issue should be addressed some where else,
the problem i face on the conference call is that when i introduce a party in the conference that is on PSTN trunk, the audio is good only from the user that has added the PSTN party into the conference, all the other parties cannot get and produce good audio, there is only a noisy sound.
So here the example :

User A call User B, then after call estabilished User A add a PSTN user into the conference on Mobile phone or landline. The 3 party are in conference but only User A and PSTN has good Audio, the User B receive only noisy audio.
The same test can be done with User B add the PSTN number into the conference, and in this case User B and PSTN are fine and User A got noisy audio.

hope some one could reproduce the same issue.

Could we see some traces? The ones in SC's log folder should be ok too.

Emil

···

On Wed, Dec 8, 2010 at 9:26 PM, Fabio Telme <fabio@telme.sg> wrote:

Regards
Fabio

Il giorno 08/dic/2010, alle ore 19.32, Emil Ivov ha scritto:

Hey Fabio,

На 08.12.10 16:38, Fabio Telme написа:

Hi

I would like to report an issue on the conference call using SIP,
The User A make a SIP call to user B, then after the call is
established negotiating the codec g722 or Speex, the User A add a
PSTN Callee into the the call. The User A and PSTN GW negotiate a
new codec (PCMA or PCMU) but no re-negotiation is sent to User B
that stay into the conference with g722 or Speex producing
"Martians" audio. When the User A ,that is the one added a new
party into the call, renegotiate the codec, should then re-invite
all the other parties to renegotiate the new codec.

Why would you want to do that?

SIP Communicator handles the transcoding.

This way whoever supports wideband can keep on using it and benefit from
good quality, and for everyone else it's just the same.

Can you explain your thoughts?

Emil

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--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
SIP Communicator
emcho@sip-communicator.org PHONE: +33.1.77.62.43.30
http://sip-communicator.org FAX: +33.1.77.62.47.31


#5

Hi Emil,

in attach 2 logs, the User A that add the PSTN into the conference with good audio stream and User B that is the user that receive and produce noisy audio.

Thanks
Fabio

sip-communicator0.log.0_userB (55.5 KB)

sip-communicator0.log_userA.0 (56 KB)


#6

Hey Fabio,

На 09.12.10 14:14, Fabio Telme написа:

Hi Emil,

in attach 2 logs, the User A that add the PSTN into the conference
with good audio stream and User B that is the user that receive and
produce noisy audio.

I am sorry, I forgot to mention: could you please enable packet logging
for all protocols and send me the entire content of the log folder? The
pcap files are of a particular interest.

Also can you describe the noise you are seeing at User B's side? You can
try recording a test call and sending it with the logs if the problem
appears in the recording as well.

Cheers

Thanks Fabio

Hi Emil,

Thanks for the answer, i did not know that SC is able to
transcode, that's really good, so i think the issue should be
addressed some where else, the problem i face on the conference
call is that when i introduce a party in the conference that is
on PSTN trunk, the audio is good only from the user that has
added the PSTN party into the conference, all the other parties
cannot get and produce good audio, there is only a noisy sound.
So here the example :

User A call User B, then after call estabilished User A add a
PSTN user into the conference on Mobile phone or landline. The 3
party are in conference but only User A and PSTN has good Audio,
the User B receive only noisy audio. The same test can be done
with User B add the PSTN number into the conference, and in this
case User B and PSTN are fine and User A got noisy audio.

hope some one could reproduce the same issue.

Could we see some traces? The ones in SC's log folder should be ok
too.

Emil

Regards Fabio

Hey Fabio,

На 08.12.10 16:38, Fabio Telme написа:

Hi

I would like to report an issue on the conference call using
SIP, The User A make a SIP call to user B, then after the
call is established negotiating the codec g722 or Speex, the
User A add a PSTN Callee into the the call. The User A
and PSTN GW negotiate a new codec (PCMA or PCMU) but no
re-negotiation is sent to User B that stay into the
conference with g722 or Speex producing "Martians" audio.
When the User A ,that is the one added a new party into the
call, renegotiate the codec, should then re-invite all the
other parties to renegotiate the new codec.

Why would you want to do that?

SIP Communicator handles the transcoding.

This way whoever supports wideband can keep on using it and
benefit from good quality, and for everyone else it's just the
same.

Can you explain your thoughts?

Emil

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dev-help@sip-communicator.dev.java.net

-- Emil Ivov, Ph.D. 67000
Strasbourg, Project Lead France
SIP Communicator emcho@sip-communicator.org
PHONE: +33.1.77.62.43.30 http://sip-communicator.org
FAX: +33.1.77.62.47.31

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···

Il giorno 09/dic/2010, alle ore 00.40, Emil Ivov ha scritto:

On Wed, Dec 8, 2010 at 9:26 PM, Fabio Telme <fabio@telme.sg> >> wrote:

Il giorno 08/dic/2010, alle ore 19.32, Emil Ivov ha scritto:

For additional commands, e-mail:
dev-help@sip-communicator.dev.java.net

--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
SIP Communicator
emcho@sip-communicator.org PHONE: +33.1.77.62.43.30
http://sip-communicator.org FAX: +33.1.77.62.47.31

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#7

Hi Emil,

i will make a pcap of the packets, the noisy is a voice that is coded and encoded wrongly, so there is a distorced voice, but with no way to recognize as any kind of voice content or meaning. I will attach recording of the sound too.

Thanks
Fabio

···

Il giorno 09/dic/2010, alle ore 14.58, Emil Ivov ha scritto:

Hey Fabio,

На 09.12.10 14:14, Fabio Telme написа:

Hi Emil,

in attach 2 logs, the User A that add the PSTN into the conference
with good audio stream and User B that is the user that receive and
produce noisy audio.

I am sorry, I forgot to mention: could you please enable packet logging
for all protocols and send me the entire content of the log folder? The
pcap files are of a particular interest.

Also can you describe the noise you are seeing at User B's side? You can
try recording a test call and sending it with the logs if the problem
appears in the recording as well.

Cheers

Thanks Fabio

Il giorno 09/dic/2010, alle ore 00.40, Emil Ivov ha scritto:

On Wed, Dec 8, 2010 at 9:26 PM, Fabio Telme <fabio@telme.sg> >>> wrote:

Hi Emil,

Thanks for the answer, i did not know that SC is able to
transcode, that's really good, so i think the issue should be
addressed some where else, the problem i face on the conference
call is that when i introduce a party in the conference that is
on PSTN trunk, the audio is good only from the user that has
added the PSTN party into the conference, all the other parties
cannot get and produce good audio, there is only a noisy sound.
So here the example :

User A call User B, then after call estabilished User A add a
PSTN user into the conference on Mobile phone or landline. The 3
party are in conference but only User A and PSTN has good Audio,
the User B receive only noisy audio. The same test can be done
with User B add the PSTN number into the conference, and in this
case User B and PSTN are fine and User A got noisy audio.

hope some one could reproduce the same issue.

Could we see some traces? The ones in SC's log folder should be ok
too.

Emil

Regards Fabio

Il giorno 08/dic/2010, alle ore 19.32, Emil Ivov ha scritto:

Hey Fabio,

На 08.12.10 16:38, Fabio Telme написа:

Hi

I would like to report an issue on the conference call using
SIP, The User A make a SIP call to user B, then after the
call is established negotiating the codec g722 or Speex, the
User A add a PSTN Callee into the the call. The User A
and PSTN GW negotiate a new codec (PCMA or PCMU) but no
re-negotiation is sent to User B that stay into the
conference with g722 or Speex producing "Martians" audio.
When the User A ,that is the one added a new party into the
call, renegotiate the codec, should then re-invite all the
other parties to renegotiate the new codec.

Why would you want to do that?

SIP Communicator handles the transcoding.

This way whoever supports wideband can keep on using it and
benefit from good quality, and for everyone else it's just the
same.

Can you explain your thoughts?

Emil

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---------------------------------------------------------------------

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dev-help@sip-communicator.dev.java.net

-- Emil Ivov, Ph.D. 67000
Strasbourg, Project Lead France
SIP Communicator emcho@sip-communicator.org
PHONE: +33.1.77.62.43.30 http://sip-communicator.org
FAX: +33.1.77.62.47.31

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dev-help@sip-communicator.dev.java.net

---------------------------------------------------------------------

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For additional commands, e-mail:
dev-help@sip-communicator.dev.java.net

--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
SIP Communicator
emcho@sip-communicator.org PHONE: +33.1.77.62.43.30
http://sip-communicator.org FAX: +33.1.77.62.47.31

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#8

На 09.12.10 15:04, Fabio Telme написа:

Hi Emil,

i will make a pcap of the packets,

You don't need to make them. Just enable packet logging for all
protocols in the "Advanced" configuration panel and SC will generate
them for you.

Of course if you prefer to use Wireshark yourself, that's also fine (as
long as u r not using TLS)

Emil

the noisy is a voice that is
coded and encoded wrongly, so there is a distorced voice, but with
no way to recognize as any kind of voice content or meaning. I will
attach recording of the sound too.

Thanks Fabio

Hey Fabio,

На 09.12.10 14:14, Fabio Telme написа:

Hi Emil,

in attach 2 logs, the User A that add the PSTN into the
conference with good audio stream and User B that is the user
that receive and produce noisy audio.

I am sorry, I forgot to mention: could you please enable packet
logging for all protocols and send me the entire content of the log
folder? The pcap files are of a particular interest.

Also can you describe the noise you are seeing at User B's side?
You can try recording a test call and sending it with the logs if
the problem appears in the recording as well.

Cheers

Thanks Fabio

Hi Emil,

Thanks for the answer, i did not know that SC is able to
transcode, that's really good, so i think the issue should
be addressed some where else, the problem i face on the
conference call is that when i introduce a party in the
conference that is on PSTN trunk, the audio is good only from
the user that has added the PSTN party into the conference,
all the other parties cannot get and produce good audio,
there is only a noisy sound. So here the example :

User A call User B, then after call estabilished User A
add a PSTN user into the conference on Mobile phone or
landline. The 3 party are in conference but only User A and
PSTN has good Audio, the User B receive only noisy audio.
The same test can be done with User B add the PSTN number
into the conference, and in this case User B and PSTN are
fine and User A got noisy audio.

hope some one could reproduce the same issue.

Could we see some traces? The ones in SC's log folder should be
ok too.

Emil

Regards Fabio

Hey Fabio,

На 08.12.10 16:38, Fabio Telme написа:

Hi

I would like to report an issue on the conference call
using SIP, The User A make a SIP call to user B, then
after the call is established negotiating the codec g722
or Speex, the User A add a PSTN Callee into the the
call. The User A and PSTN GW negotiate a new codec
(PCMA or PCMU) but no re-negotiation is sent to User B
that stay into the conference with g722 or Speex
producing "Martians" audio. When the User A ,that is the
one added a new party into the call, renegotiate the
codec, should then re-invite all the other parties to
renegotiate the new codec.

Why would you want to do that?

SIP Communicator handles the transcoding.

This way whoever supports wideband can keep on using it
and benefit from good quality, and for everyone else it's
just the same.

Can you explain your thoughts?

Emil

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dev-help@sip-communicator.dev.java.net

-- Emil Ivov, Ph.D. 67000
Strasbourg, Project Lead
France SIP Communicator emcho@sip-communicator.org PHONE:
+33.1.77.62.43.30 http://sip-communicator.org FAX:
+33.1.77.62.47.31

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-- Emil Ivov, Ph.D. 67000
Strasbourg, Project Lead France
SIP Communicator emcho@sip-communicator.org
PHONE: +33.1.77.62.43.30 http://sip-communicator.org
FAX: +33.1.77.62.47.31

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···

Il giorno 09/dic/2010, alle ore 14.58, Emil Ivov ha scritto:

Il giorno 09/dic/2010, alle ore 00.40, Emil Ivov ha scritto:

On Wed, Dec 8, 2010 at 9:26 PM, Fabio Telme <fabio@telme.sg> >>>> wrote:

Il giorno 08/dic/2010, alle ore 19.32, Emil Ivov ha scritto:

For additional commands, e-mail:
dev-help@sip-communicator.dev.java.net

--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
SIP Communicator
emcho@sip-communicator.org PHONE: +33.1.77.62.43.30
http://sip-communicator.org FAX: +33.1.77.62.47.31

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